Dear
I am using this function with L
for example in the dbase.
app=Dial
appdata=SIP/[EMAIL PROTECTED]|60|L(10)
it means dial 1 thru 1.1.1.1, with
limitation=10 mili-second, and time out=60 sec
best
Mani
--- Bhrugu Mehta [EMAIL PROTECTED] wrote:
hi, all
proble:
I have add
I've used http://www.555-1212.com, but not at the volume you're talking
about. Maybe you can work a deal with them.
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: [EMAIL PROTECTED]
We've started testing Asterisk 1.4 1.2 has been very stable and we
have processed millions of minutes with it, SIP-to-ZAP, SIP-to-SIP and
SIP-to-IAX. We've been using Asterisk over 4 years now and it has
really re-invented the way me and a few others think of telephones.
The only inter-op
22 dec 2007 kl. 10.55 skrev Andrew Joakimsen:
We've started testing Asterisk 1.4 1.2 has been very stable and we
have processed millions of minutes with it, SIP-to-ZAP, SIP-to-SIP and
SIP-to-IAX. We've been using Asterisk over 4 years now and it has
really re-invented the way me and a
We expect Kerry Garrison to respond to this live Friday 21st Dec at 12
Noon EST with what steps they are taking and why.
http://VoipUsersConference.org
IRC: #voip-users-conference on Freenode.net
Thanks to all who participated in the call. A lot of interesting side
issues came up such as who
Take a look at www.411xml.com
On Dec 19, 2007 4:35 PM, Norman Franke [EMAIL PROTECTED] wrote:
Is anyone aware of a service where we can lookup phone numbers to
determine a name and/or name + address available in bulk?
We want to look up every number called to our call center, so it will
be
On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:
Olle E Johansson [EMAIL PROTECTED] writes:
But on the other hand, if people rely on third-party distributions
we might want to set up some kind of peer pressure on the
maintainers - and possibly identify them so we can support
For the price Grandstream GXP-2000 is very feature packed and has a
decent size and resolution display. The menus aren't the nicest but
the phone works and it does not sound bad. For $70 you get what you
pay for and the firmware is pretty stable and always being updated.
On Dec 19, 2007 11:33 PM,
http://spc.pifiu.com for the stuff Linksys are Nazis about.
On Dec 21, 2007 1:56 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote:
Hi!
d tbsky wrote:
ok. i will add linksys to our testing list. but cisco tend to lock things.
can we get firmware for linksys easily ? or we must pay like cisco
Hi,
in the earlier version there was a sounds.txt with the transcript of the
soundfiles. Does this still exist somewhere?
Is there a plan to make speech synthesis available the same way as
soundfiles, ie. instead of playing language/soundfile.wav, send the text to
the speechengine and play the
Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy
when either of 200 or 201 are in use?
exten = 200,hint,SIP/200SIP/201
exten = 201,hint,SIP/200SIP/201
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit
On Sat, 22 Dec 2007, Andrew Joakimsen wrote:
For the price Grandstream GXP-2000 is very feature packed and has a
decent size and resolution display. The menus aren't the nicest but
the phone works and it does not sound bad. For $70 you get what you
pay for and the firmware is pretty stable
On Saturday 22 December 2007 01:51:56 am Johansson Olle E wrote:
With that, I'm now changing my focus from SIP invite states,
RTP sessions and video formats to Christmas ham purchasing,
baking Christmas bread (julvört) and decorating the Christmas
tree. Of course, you understand that there's
You're right of course. I should have dug into this a little deeper
and checked to see if it is corrected in the current release. As is
so often the case, I was working on a real specific problem and once
the system started doing what I wanted it to I pretty much forgot
about it.
I
Hi,
The message that asterisk receives is not a retransmission but this is the
same message but it enters asterisk from other sip proxy which is not a
loop.
The flow is the following
Asterisk SIP Proxy (Location Service)
INVITE (to registrar)
-
INVITE (to voicemail when
Hi!
Now over to a summary of the feedback. I'm not going deeper into bugs
reported, those will be handled separately.
Looks like I am a bit late, but I'll try to add my share as well to
highlight some of the issues that are invovled with 1.2 to 1.4
transition:
- with the advent of the
22 dec 2007 kl. 15.51 skrev Tomasz Zieleniewski:
Hi,
The message that asterisk receives is not a retransmission but this
is the same message but it enters asterisk from other sip proxy
which is not a loop.
The flow is the following
Asterisk SIP Proxy (Location Service)
INVITE
call-limit is to set number of alternate calls . and L is to limit
duration of each call .
On Dec 22, 2007 2:54 PM, Pezhman Lali [EMAIL PROTECTED] wrote:
Dear
I am using this function with L
for example in the dbase.
app=Dial
appdata=SIP/[EMAIL PROTECTED]|60|L(10)
it means dial 1
At 01:51 12/22/2007, Johansson Olle E wrote:
Friends,
We might have to reconsider our support policy here, where we
developers abandoned 1.2 this summer. We might need another
team that runs 1.2 support in the bug tracker.
Pretty please, with cranberry sauce on top.
With that, I'm now changing my focus from SIP invite states, RTP
sessions and video formats to Christmas ham purchasing, baking
Christmas bread (julvört) and decorating the Christmas tree. Of
course, you understand that there's an Asterisk asterisk on top of
all those trees, right? :-)
Andrew Joakimsen wrote:
{emphasis added}What are the plans for Asterisk 1.6 in regards to
furthering T.38 support?{/emphasis added}
If you really want further T.38 support, then you should be looking at
callweaver. (An Asterisk 1.2 branch).
The T.38 support appears to be a lot better than
You can not do this. You can not have an INVITE that Asterisk originated
enter back into Asterisk. Technically this is not a loop, but this is an
INVITE glare and the way Asterisk is reacting is correct.
You'll need to change the Call-Id of the INVITE that goes into Asterisk (a
proxy can not do
On Thu, 13 Dec 2007 20:40:08 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
One of the major advantages of using voip is that call termination and
DIDs are wholly separate matters. You can send outbound calls to
various ITSPs based on least cost routing, leaving your POTS lines free
to take
Hello
Since I got the IBM Netvista to boot Linux, and am still waiting for
the Compact Flash cards that I ordered, I was wondering if someone
knew of an Asterisk distribution that can run on that kind of diskless
host?
I've taken a look at AstLinux and AskoziaPBX, but they both seem to be
meant
On Sun, 23 Dec 2007 02:29:13 +0100, Vincent wrote:
On Thu, 13 Dec 2007 20:40:08 -0600, Michael Graves
[EMAIL PROTECTED] wrote:
One of the major advantages of using voip is that call termination and
DIDs are wholly separate matters. You can send outbound calls to
various ITSPs based on least
On Sun, 23 Dec 2007 02:34:45 +0100, Vincent wrote:
Hello
Since I got the IBM Netvista to boot Linux, and am still waiting for
the Compact Flash cards that I ordered, I was wondering if someone
knew of an Asterisk distribution that can run on that kind of diskless
host?
I've taken a look at
On Sun, Dec 23, 2007 at 02:34:45AM +0100, Vincent wrote:
Hello
Since I got the IBM Netvista to boot Linux, and am still waiting for
the Compact Flash cards that I ordered, I was wondering if someone
knew of an Asterisk distribution that can run on that kind of diskless
host?
Yes. I have a
Depending on how many faxes you have coming in a simple fxs/fxo card
will do the trick .. either Sagnoma or Digium or any others you could
also use any decent ATA.. Asterisk only needs to know its a fax and what
dialed number it came on to route it to the correct fax machine.
Asterisk would
23 dec 2007 kl. 01.45 skrev Raj Jain:
You can not do this. You can not have an INVITE that Asterisk
originated enter back into Asterisk. Technically this is not a loop,
but this is an INVITE glare and the way Asterisk is reacting is
correct.
You'll need to change the Call-Id of the
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